Device and method for transmitting and receiving voice data in wireless communication system

ABSTRACT

Provided are a device and a method for transmitting and receiving voice data in a wireless communication system. A method for operating a transmission terminal for transmitting a voice signal comprises the steps of: generating sampling and bitrate request information including sampling information for determining a sampling rate of the voice signal and bitrate information for determining a bitrate of the voice signal, and transmitting the generated sampling and bitrate request information to a reception terminal; receiving, from the reception terminal, combined determination information obtained by at least one combination of the sampling rate determined on the basis of the sampling information and the bitrate determined on the basis of the bitrate information; and compressing the voice signal according to the received combined determination information, and transmitting the compressed voice signal to the reception terminal.

CROSS-REFERENCE TO RELATED APPLICATION(S)

This application is a Continuation of U.S. application Ser. No.15/305,944, filed on Mar. 20, 2017, which is a National Stage ofInternational Application No. PCT/KR2015/006330 filed on Jun. 22, 2015,which claims priority to Korean Patent Application No. 10-2014-0153191,filed on Nov. 5, 2014, in the Korean Intellectual Property Office, theentire disclosures of which are incorporated herein by reference for allpurposes.

TECHNICAL FIELD

Exemplary embodiments relate to an apparatus and method for transmittingand receiving voice data in a wireless communication system.

BACKGROUND ART

A speech is compressed by a speech codec (encoder) of a transmissionterminal, various types of headers such as real-time transport protocol(RTP), user datagram protocol (UDP), and Internet protocol (IP) headersare attached to the compressed speech, and the header-attachedcompressed speech is transmitted through a communication modem such as along-term evolution (LTE) modem. A reception terminal sequentiallyremoves the RTP, UDP, and IP headers and checks whether there is a lostspeech frame. A speech frame restored without any loss is decompressedby a speech codec (decoder) from a compressed state, converted into apulse coded modulation (PCM) signal, and delivered to a speaker.

When a loss in information is discovered during the restoration ofspeech frames, error concealment for reducing damages in sound qualityis carried out by using information about previous frames. If a call isnot smooth because speech frames were lost during transmission due tocongestion of a transmission path, a speech bit-rate may be adjusted bytransmitting a codec mode request (CMR) message to a counterpartterminal such that the bit-rate is temporarily decreased when there is acongestion status and is gradually increased as the congestion status issolved.

According to conventional voice compression techniques such as adaptivemulti-rate (AMR) and adaptive multi-rate wideband (AMR-WB) techniques, avoice bandwidth is fixed to a narrowband or a wideband, this indicatesthat an analog voice signal is converted into a digital signal at 8,000or 16,000 samples/s and compressed, and this conversion speed does notchange during a call. An AMR codec may compress a voice signal digitizedat 8,000 samples/s to eight types of bit-rates of 4.75 to 12.2 Kbps andprocess a voice signal of a band of 300 to 3,400 Hz. An AMR-WB codec maycompress a voice signal digitized at 16,000 samples/s to nine types ofbit-rates of 6.6 to 23.85 Kbps and process a voice signal of a band of50 to 7,000 Hz.

Recently, with respect to call quality of AMR-WB voice codecscommercialized in a voice over LTE (VoLTE) service, most listeners thinkthat call quality is improved, but some listeners do not prefer callquality of a high-frequency voice and tend to think that the quality ofa conventional AMR voice codec was better. This indicates that ahigh-frequency voice component may or may not be preferred according toa listener or background noise compressed along with a voice.

DETAILED DESCRIPTION OF THE INVENTION Technical Problem

Provided are an operating method and apparatus of a transmissionterminal and a reception terminal for transmitting/receiving a voicesignal of which a sampling rate and/or a bit-rate are mutually adjustedbetween terminals during a call in consideration of the fact thatoptimal call quality can be achieved by using another sampling rateaccording to circumstances even at the same voice bit-rate.

Provided are a method and apparatus for efficiently negotiating, betweentwo terminals, a bit-rate and/or a sampling rate of a voice codecsupporting bit-rates of a wider range than the prior art and samplingrates of a wider range than the prior art.

Technical Solution

According to an aspect of an exemplary embodiment, an operating methodof a transmission terminal for transmitting a voice signal includes:generating sampling- and bit-rate request information including samplinginformation for determining a sampling rate of a voice signal andbit-rate information for determining a bit-rate of the voice signal andtransmitting the generated sampling- and bit-rate request information toa reception terminal; receiving, from the reception terminal,combination determination information by at least one combination ofsampling rates determined based on the sampling information andbit-rates determined based on the bit-rate information; and compressingthe voice signal according to the received combination determinationinformation and transmitting the compressed voice signal to thereception terminal.

According to an aspect of another exemplary embodiment, an operatingapparatus of a transmission terminal for transmitting a voice signalincludes: a request information generation unit configured to generatesampling- and bit-rate request information including samplinginformation for determining a sampling rate of a voice signal andbit-rate information for determining a bit-rate of the voice signal; acontrol unit configured to control the generated sampling- and bit-raterequest information to be transmitted to a reception terminal; aninterface unit configured to transmit the sampling- and bit-rate requestinformation to the reception terminal under control of the control unit;and a voice compression unit configured to, if the interface unitreceives, from the reception terminal, combination determinationinformation by at least one combination of sampling rates determinedbased on the sampling information and bit-rates determined based on thebit-rate information, compress the voice signal according to thereceived combination determination information, wherein the interfaceunit transmits the compressed voice signal to the reception terminalunder control of the control unit.

According to an aspect of another exemplary embodiment, an operatingmethod of a reception terminal for receiving a voice signal includes:receiving, from a transmission terminal, sampling- and bit-rate requestinformation including sampling information for determining a samplingrate of a voice signal and bit-rate information for determining abit-rate of the voice signal; determining at least one combination ofsampling rates determined based on the sampling information andbit-rates determined based on the bit-rate information, according to thesampling- and bit-rate request information; and transmitting thedetermined combination determination information to the transmissionterminal.

According to an aspect of another embodiment, an operating apparatus ofa reception terminal for receiving a voice signal includes: an interfaceunit configured to receive, from a transmission terminal, sampling- andbit-rate request information including sampling information fordetermining a sampling rate of a voice signal and bit-rate informationfor determining a bit-rate of the voice signal; a combinationdetermination unit configured to determine at least one combination ofsampling rates determined based on the sampling information andbit-rates determined based on the bit-rate information, according to thesampling- and bit-rate request information; and a control unitconfigured to control the determined combination determinationinformation to be transmitted to the transmission terminal, wherein theinterface unit transmits the combination determination information tothe transmission terminal under control of the control unit.

Advantageous Effects of the Invention

According to exemplary embodiments, in a voice over Internet protocol(VoIP) system using a voice codec such as enhanced voice services (EVS)capable of compressing a voice at a plurality of sampling rates andbit-rates, a sampling rate and a bit-rate to be used for a service inthe voice codec supporting the plurality of sampling rates and bit-ratesmay be appropriately negotiated, and a compression scheme of acounterpart terminal may be dynamically adjusted according to a taste ofa recipient, voice content, and background noise.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a flowchart of an operating method of a transmission terminalfor transmitting a voice signal, according to an exemplary embodiment.

FIG. 2 is a reference diagram showing an example of a payload header.

FIG. 3 is a reference diagram showing an example of a VoIP packetincluding a payload header.

FIGS. 4A and 4B are reference diagrams showing an example of informationtransmitted from a transmission terminal to a reception terminal andinformation transmitted from the reception terminal to the transmissionterminal.

FIGS. 5A and 5B are reference diagrams for describing a receptionterminal determining combination determination information.

FIG. 6 is a block diagram of an operating apparatus of a transmissionterminal for transmitting a voice signal, according to an exemplaryembodiment.

FIG. 7 is a flowchart of an operating method of a reception terminal forreceiving a voice signal, according to an exemplary embodiment.

FIG. 8 is a block diagram of an operating apparatus of a receptionterminal for receiving a voice signal, according to an exemplaryembodiment.

FIG. 9 is an illustrative reference diagram for describing two terminalsnegotiating a bit-rate to be used for a call in a range unit.

FIG. 10 is an illustrative reference diagram for describing twoterminals negotiating a sampling rate to be used for a call in a rangeunit.

FIG. 11 is a reference diagram showing an example of informationtransmitted from a transmission terminal to a reception terminal andinformation transmitted from the reception terminal to the transmissionterminal between the two terminals which negotiate a bit-rate in a rangeunit.

FIG. 12 is a reference diagram showing an example of informationtransmitted from a transmission terminal to a reception terminal andinformation transmitted from the reception terminal to the transmissionterminal between the two terminals which negotiate a sampling rate in arange unit.

FIG. 13 is a reference diagram showing an example of informationtransmitted from a transmission terminal to a reception terminal andinformation transmitted from the reception terminal to the transmissionterminal between the two terminals which negotiate a bit-rate and asampling rate in a range unit.

FIG. 14 is a reference diagram showing an example of informationtransmitted from a transmission terminal to a reception terminal andinformation transmitted from the reception terminal to the transmissionterminal between the two terminals which negotiate a bit-rate and asampling rate in a range unit.

MODE OF THE INVENTION

FIGS. 1 to 14 are only illustrative and thus should not be analyzed asbeing limitations of the scope of the invention. It will be understoodby those of ordinary skill in the art that the present disclosure couldbe implemented even in a random communication system includingappropriate operational principles related to embodiments.

FIG. 1 is a flowchart of an operating method of a transmission terminalfor transmitting a voice signal, according to an exemplary embodiment.

Referring to FIG. 1, in operation S100, the transmission terminal maygenerate sampling- and bit-rate request information including samplinginformation for determining a sampling rate and bit-rate information fordetermining a bit-rate of a voice signal and transmit the generatedsampling- and bit-rate request information to a reception terminal. Thesampling information may include sampling list information having atleast one sampling type or sampling rate range information determinedwith respect to a specific range, and the bit-rate information mayinclude bit-rate list information having at least one bit-rate type orbit-rate range information determined with respect to a specific range.

In general, examples of types of voice codecs according to bands are asshown in Table 1.

TABLE 1 Source Bandwidth Sampling Rate Bit-rate Codec Type (Hz)(samples/s) (Kbps) Narrow Band (NB) 100~3,500  8,000 5.9, 7.2, 8, 9.6,13.2, 16.4, 24.4, 32, 48, 64, 96, 128 Wide Band (WB) 50~7,000  16,000Super Wide Band 50~16,000 32,000 (SWB) Full Band (FB) 50~20,000 48,000

According to Table 1, a voice codec of the NB corresponding to 100 to3,500 Hz has a sampling rate of 8,000 samples/s and selectively uses, asa bit-rate, one of 7.2, 8, 9.6, 13.2, 16.4, 24.4, 32, 48, 64, 96, and128. In addition, a voice codec of the WB corresponding to 50 to 7,000Hz has a sampling rate of 16,000 samples/s and selectively uses one ofthe 11 bit-rates described above. In addition, a voice codec of the SWBcorresponding to 50 to 16,000 Hz has a sampling rate of 32,000 samples/sand selectively uses one of the 11 bit-rates described above. Inaddition, a voice codec of the FB corresponding to 50 to 20,000 Hz has asampling rate of 48,000 samples/s and selectively uses one of the 11bit-rates described above. That is, the voice codecs according to bandsselectively use various bit-rates but respectively use fixed samplingrates. Compared with this, the exemplary embodiment allows not onlybit-rates but also sampling rates to be selectively used according totypes of voice codecs as described below.

According to the voice compression techniques, the number of bit-ratesusable during a call is 8, 9, or less, and a difference between aminimum value and a maximum value is not that large as 10-20 Kbps orless. For example, 8 bit-rates between 4.75 Kbps and 12.2 Kbps may beused in the AMR technique, and 9 bit-rates between 6.6 Kbps and 23.85Kbps may be used in the AMR-WB technique.

According to an exemplary embodiment, the transmission terminal maygenerate list information of, for example, 8,000, 16,000, 32,000, and48,000 samples/s corresponding to types of sampling rates as samplinglist information corresponding to sampling and/or bit-rate requestinformation. In addition, the transmission terminal may generate listinformation of, for example, 7.2, 8, 9.6, 13.2, 16.4, 24.4, 32, 48, 64,96, and 128 as bit-rate list information.

The sampling list information included in the sampling- and bit-raterequest information may include sampling identification informationcorresponding to each of sampling types, and the bit-rate listinformation may include bit-rate identification informationcorresponding to each bit-rate type.

Table 2 illustrates identification information according to samplingrates.

TABLE 2 Sampling Rate (samples/s) 8,000 16,000 32,000 48,000 Id 0 1 2 3

According to Table 2, identification information Id is allocated to eachof sampling rates.

Table 3 illustrates identification information according to bit-rates.

TABLE 3 Bit-rate (Kbps) 7.2 8 9.6 13.2 16.4 24.4 32 48 64 96 128 Id 0 12 3 4 5 6 7 8 9 10

According to Table 3, identification information Id is allocated to eachbit-rate.

The transmission terminal may insert the generated sampling- andbit-rate request information into a payload header. That is, thesampling list information and the bit-rate list informationcorresponding to the sampling- and bit-rate request information may beadded to the payload header.

The payload header includes, together with the sampling list informationand the bit-rate list information, sampling type confirmationinformation of a sampling type being currently used, bit-rate typeconfirmation information of a bit-rate type being currently used, andthe like.

FIG. 2 is a reference diagram showing an example of a payload header. InFIG. 2, “Codec Mode Request” indicates bit-rate list information, “BWRequest” indicates sampling list information, “Codec Mode Used”indicates bit-rate type confirmation information of a bit-rate typebeing currently used, and “BW Used” indicates sampling type confirmationinformation of a sampling type being currently used. In addition, “F”indicates information indicating whether a voice frame is continuouslytransmitted, and “0” indicates information indicating whether there isan error in a voice frame.

A function of each field of the payload header of FIG. 2 may beillustrated as Table 4 below. As in an enhanced voice service (EVS) formultiple bit-rates/multiple bandwidths, a payload header of a voicecodec includes 2 bytes, and a function and a length of each field are asfollows.

TABLE 4 Each Length Field (Bits) Function Codec 4 bit-rate requested tocounterpart terminal, 0-10 can Mode be assigned with 0000-1010: 7.2 (0),8 (1), 9.6 (2), Request 13.2 (3), 16.4 (4), 24.4 (5), 32 (6), 48 (7), 64(8), 96 (9), 128 (10) BW 2 bandwidth requested to counterpart terminal:00 Request (NB, 8000 samples/s), 01 (WB, 16000 samples/s), 10 (SWB,32000 samples/s), 11 (FB, 48000 samples/s) F 1 F = 1: voice frame iscarried after corresponding frame, F = 0: corresponding frame is lastvoice frame Codec 4 bit-rate used for current frame, 0-10 can beassigned Mode with 0000-1010: 7.2 (0), 8 (1), 9.6 (2), 13.2 (3), Used16.4 (4), 24.4 (5), 32 (6), 48 (7), 64 (8), 96 (9), 128 (10) BW Used 2bandwidth used for current frame: 00 (NB, 8000 samples/s), 01 (WB, 16000samples/s), 10 (SWB, 32000 samples/s), 11 (FB, 48000 samples/s) Q 1 F =0: corresponding frame includes error, F = 0: corresponding frameincludes no error Reserved 2 Not used

That is, as the bit-rate list information, list information of bit-ratetypes, for example, 7.2, 8, 9.6, 13.2, 16.4, 24.4, 32, 48, 64, 96, and128 and respective pieces of identification information 0000 to 1010corresponding to the list information are included. In addition, as thesampling rate list information, list information of, for example, 8,000,16,000, 32,000, and 48,000 and respective pieces of identificationinformation 00 to 11 corresponding to the list information are included.In addition, as the bit-rate type confirmation information, listinformation of a bit-rate used for a current frame and identificationinformation corresponding to the list information are included. Inaddition, as the sampling type confirmation information, informationabout a sampling rate used for the current frame and identificationinformation corresponding to the information are included.

The transmission terminal may generate a voice over Internet protocol(VoIP) packet in which a real-time transport protocol (RTP) header, auser datagram protocol (UDP) header, and an Internet protocol (IP)header are sequentially added in addition to a payload header having thesampling- and bit-rate request information, and transmit the generatedVoIP packet to the reception terminal.

FIG. 3 is a reference diagram showing an example of a VoIP packetincluding a payload header. A payload header is added to a voice frame,wherein “0” may be padded to adjust a payload length in a byte unit. Asdescribed above, sampling rate list information and bit-rate listinformation are included in the payload header, and sampling rateinformation and bit-rate information being currently used may beincluded. In this case, a sum of the payload header, an EVS frame, andpadding bits is named an RTP payload. A VoIP packet having an RTPheader, a UDP header, and an IP header in addition to the RTP payloadmay be transmitted to a counterpart terminal through a modem.

Referring back to FIG. 1, after operation S100, in operation S102, thetransmission terminal may receive, from the reception terminal,combination determination information by at least one combination ofsampling rates determined based on the sampling information andbit-rates determined based on the bit-rate information. The combinationdetermination information may be determined by at least one combinationof sampling types selected from the sampling information and bit-ratetypes selected from the bit-rate information or by at least onecombination of a range of sampling rates determined within the range ofthe sampling rate range information and a range of bit-rates determinedwithin the range of the bit-rate range information. The sampling raterange information may be differently determined in a sending directionand a receiving direction, and the bit-rate range information may bedifferently determined in the sending direction and the receivingdirection.

FIG. 4 is a reference diagram showing an example of informationtransmitted from a transmission terminal to a reception terminal andinformation transmitted from the reception terminal to the transmissionterminal. FIG. 4(a) shows information for informing the receptionterminal that the transmission terminal can use three types of voicecodes of EVS, AMR-WB, and AMR for voice communication. The informationindicates that EVS supports up to the FB (48,000 samples/s), AMR-WBsupports 16,000 samples/s, and AMR supports 8,000 samples/s. That is,EVS/48000, AMR-WB/16000, and AMR/8000 indicate list information ofsampling rate types which can be combined by the reception terminal. Inaddition, b=AS:160 indicates an application-specific maximum bit-rateand has an integer value obtained by adding an RTP/UDP/IP header to amaximum voice bit-rate. That is, b=AS:160 indicates a value obtained byadding 32 (the RTP/UDP/IP header) to 128 Kbps of EVS. Therefore, theapplication-specific maximum bit-rate is a maximum value of bit-ratesand indicates list information of bit-rate types which can be combinedby the reception terminal.

When receiving sampling- and bit-rate list information includingsampling rate list information and bit-rate list information from thetransmission terminal, the reception terminal selects EVS from thesampling rate list information and determines any one sampling rate (forexample, 32,000 samples/s) as shown in FIG. 4(b). In addition, thereception terminal determines that a bit-rate of maximum 24.4 Kbps fromthe bit-rate list information is used. That is, config-set indicatescombination determination information which is to be transmitted to thetransmission terminal as {(0, 0), (1, 1), (1, 2), (2, 3), (2, 4), (2,5)}. Each piece of the combination determination information indicates(sampling rate identification information, bit-rate identificationinformation). That is, (0, 0) is a combination of sampling rateidentification information “0” and bit-rate identification information“0”, wherein a sampling rate type corresponding to the identificationinformation corresponds to 8,000 (NB), and a bit-rate type correspondingto the identification information corresponds to 7.2. Therefore, {(0,0), (1, 1), (1, 2), (2, 3), (2, 4), (2, 5)} as the combinationdetermination information respectively indicate combination informationof sampling rate types and bit-rate types of 7.2 (NB), 8 (WB), 9.6 (WB),13.2 (SWB), 16.4 (SWB), and 24.4 (SWB).

This combination determination information corresponds to combinationinformation of sampling rate types and bit-rate types determined byusing at least one of background noise, audio quality, audio tone of anaudio signal including the voice signal.

FIG. 5 is an illustrative reference diagram for describing a receptionterminal determining combination determination information. FIG. 5(a)illustrates a voice signal based on a low-frequency band, and FIG. 5(b)illustrates a voice signal based on a high-frequency band. For example,a user of the reception terminal for receiving a voice signal may notwant to hear high-frequency audio or a sharp sound. In this case, thereception terminal may analyze received voice frames and determineinformation, i.e., combination determination information, for making acounterpart terminal decrease a sampling rate from the super wideband(SWB) to the wideband (WB) or narrowband (NB) when an energy ratio ofhigh-frequency components higher than a specific frequency or higher isa specific level or higher as shown in FIG. 5(b). This situation mayoccur in the inside of a factory in which metallic noise is generated orin a case of talking to a person having a high tone. On the contrary, ina situation of being connected to a security camera or the like suchthat it is necessary to transmit even a change in a fine sound, it maybe adjusted so as to increase a sampling rate as high as possible.

Therefore, the reception terminal determines proper combinationdetermination information for a voice signal in consideration ofbackground noise, audio quality, a tone of a caller, and the like of avoice signal provided from a counterpart terminal and transmits thedetermined combination determination information to the counterpartterminal (transmission terminal), and the counterpart terminal(transmission terminal) receives the combination determinationinformation.

After operation S102, the transmission terminal may compress a voicesignal according to the received combination determination informationand transmit the compressed voice signal to the reception terminal inoperation S104. For example, when receiving {(0, 0), (1, 1), (1, 2), (2,3), (2, 4), (2, 5)} shown in FIG. 4(b) as the combination determinationinformation, the transmission terminal may compress a voice signal to betransmitted to the reception terminal, at a sampling rate and a bit-ratecorresponding to the combination determination information. That is, ifit is determined that the voice signal is compressed so as to correspondto (0, 0) of the combination determination information, the transmissionterminal may compress the voice signal according to a sampling rate type8,000 (NB) corresponding to identification information “0” and abit-rate type 7.2 corresponding to identification information “0” andtransmit the compressed voice signal to the reception terminal.

FIG. 6 is a block diagram of an operating apparatus of a transmissionterminal for transmitting a voice signal, according to an exemplaryembodiment, and the apparatus may include a request informationgeneration unit 200, a control unit 210, an interface unit 230, and avoice compression unit 240.

Referring to FIG. 6, the request information generation unit 200 maygenerate sampling- and bit-rate request information including samplinglist information having at least one sampling type for determining asampling rate of the voice signal and bit-rate list information havingat least one bit-rate type for determining a bit-rate of the voicesignal. The sampling list information included in the sampling- andbit-rate request information may include sampling identificationinformation corresponding to respective sampling types, and the bit-ratelist information may include bit-rate identification informationcorresponding to respective bit-rate types.

The request information generation unit 200 may insert the generatedsampling- and bit-rate request information into a payload header. Thatis, the sampling list information and the bit-rate list informationcorresponding to the sampling- and bit-rate request information may beadded to the payload header. The payload header includes, together withthe sampling list information and the bit-rate list information,sampling type confirmation information of a sampling type beingcurrently used, bit-rate type confirmation information of a bit-ratetype being currently used, and the like.

The request information generation unit 200 may generate a VoIP packetin which at least one of an RTP header, a UDP header, and an IP headerare added in addition to the payload header having the sampling- andbit-rate request information. As shown in FIG. 3, the requestinformation generation unit 200 inserts the sampling rate listinformation and the bit-rate list information into the payload headerand may insert sampling rate information and bit-rate information beingcurrently used into the payload header. The request informationgeneration unit 200 may generate a VoIP packet in which the RTP header,the UDP header, and the IP header are added in addition to an RTPpayload including the payload header, an EVS frame, and padding bits.

The control unit 210 may control the sampling- and bit-rate requestinformation including the sampling rate list information and thebit-rate list information to be transmitted to a reception terminal.

The interface unit 230 may transmit the sampling- and bit-rate requestinformation to the reception terminal under control of the control unit210. Thereafter, the interface unit 230 may receive, from the receptionterminal, combination determination information by at least onecombination of sampling types selected from the sampling listinformation and bit-rate types selected from the bit-rate listinformation.

For example, as shown in FIG. 4(b), {(0, 0), (1, 1), (1, 2), (2, 3), (2,4), (2, 5)} as the combination determination information may be receivedfrom the reception terminal. Each piece of the combination determinationinformation may indicate (sampling rate identification information,bit-rate identification information). Therefore, {(0, 0), (1, 1), (1,2), (2, 3), (2, 4), (2, 5)} as the combination determination informationrespectively indicate combination information of sampling rate types andbit-rate types of 7.2 (NB), 8 (WB), 9.6 (WB), 13.2 (SWB), 16.4 (SWB),and 24.4 (SWB). This combination determination information may bedetermined by using at least one of background noise, voice quality,voice tone of the voice signal.

The voice compression unit 240 may compress the voice signal accordingto the received combination determination information. For example, whenreceiving {(0, 0), (1, 1), (1, 2), (2, 3), (2, 4), (2, 5)} shown in FIG.4(b) as the combination determination information, the voice compressionunit 240 may compress a voice signal to be transmitted to the receptionterminal, at a sampling rate and a bit-rate corresponding to thecombination determination information. That is, if it is determined thatthe voice signal is compressed so as to correspond to (0, 0) of thecombination determination information, the voice compression unit 240may compress the voice signal according to a sampling rate type 8,000(NB) corresponding to identification information “0” and a bit-rate type7.2 corresponding to identification information “0” and output thecompressed voice signal to the interface unit 230.

The voice compression unit 240 may transmit the compressed voice signalto the reception terminal under control of the control unit 210.

FIG. 7 is a flowchart of an operating method of a reception terminal forreceiving a voice signal, according to an exemplary embodiment.

Referring to FIG. 7, in operation S300, the reception terminal mayreceive, from a transmission terminal, sampling- and bit-rate requestinformation including sampling information for determining a samplingrate of a voice signal and bit-rate information for determining abit-rate of the voice signal. The sampling information may includesampling list information having at least one sampling type or samplingrate range information determined to a specific range, and the bit-rateinformation may include bit-rate list information having at least onebit-rate type or bit-rate range information determined to a specificrange. The sampling list information may include sampling identificationinformation corresponding to respective sampling types, and the bit-ratelist information may include bit-rate identification informationcorresponding to respective bit-rate types.

The sampling- and bit-rate request information may be received by beinginserted into a payload header. That is, the sampling list informationand the bit-rate list information corresponding to the sampling- andbit-rate request information may be added to the payload header, and thepayload header includes, together with the sampling list information andthe bit-rate list information, sampling type confirmation information ofa sampling type being currently used, bit-rate type confirmationinformation of a bit-rate type being currently used, and the like.

The payload header having the sampling- and bit-rate request informationmay be received in a form of a VoIP packet in which at least one of anRTP header, a UDP header, and an IP header are added. As shown in FIG.3, the reception terminal may receive a VoIP packet in which the RTPheader, the UDP header, and the IP header are added in addition to anRTP payload including the payload header, an EVS frame, and paddingbits.

After operation S300, the reception terminal may determine at least onecombination of sampling rates determined based on the samplinginformation and bit-rates determined based on the bit-rate informationin operation S302. Combination determination information may bedetermined by at least one combination of sampling types selected fromthe sampling information and bit-rate types selected from the bit-rateinformation or by at least one combination of a range of sampling ratesdetermined within the range of the sampling rate range information and arange of bit-rates determined within the range of the bit-rate rangeinformation. The sampling rate range information may be differentlydetermined in a sending direction and a receiving direction, and thebit-rate range information may be differently determined in the sendingdirection and the receiving direction.

The reception terminal may separate the payload header, the RTP header,the UDP header, and the IP header from the received VoIP packet, extractthe sampling list information and the bit-rate list information from theseparated payload header, and determine the at least one combination byusing the extracted sampling list information and bit-rate listinformation.

As shown in FIG. 4(b), the reception terminal may select EVS from thesampling rate list information and determine any one sampling rate (forexample, 32,000 samples/s). In addition, the reception terminal maydetermine that a bit-rate of maximum 24.4 Kbps from the bit-rate listinformation is used. That is, config-set may be determined to be {(0,0), (1, 1), (1, 2), (2, 3), (2, 4), (2, 5)} as the combinationdetermination information. Each piece of the combination determinationinformation may indicate (sampling rate identification information,bit-rate identification information). That is, (0, 0) is a combinationof sampling rate identification information “0” and bit-rateidentification information “0”, wherein a sampling rate typecorresponding to the identification information corresponds to 8,000(NB), and a bit-rate type corresponding to the identificationinformation corresponds to 7.2. Therefore, {(0, 0), (1, 1), (1, 2), (2,3), (2, 4), (2, 5)} as the combination determination informationrespectively indicate combination information of sampling rate types andbit-rate types of 7.2 (NB), 8 (WB), 9.6 (WB), 13.2 (SWB), 16.4 (SWB),and 24.4 (SWB).

The reception terminal may determine combination information of samplingrate types and bit-rate types by using at least one of background noise,audio quality, audio tone of an audio signal including the voice signal.

As shown in FIG. 5, the reception terminal may not want to hearhigh-frequency audio or a sharp sound. In this case, the receptionterminal analyzes received voice frames and determines combinationdetermination information by including a sampling rate which is to bedecreased from the SWB to the WB or NB when an energy ratio ofhigh-frequency components of a specific frequency or higher is aspecific level or higher as shown in FIG. 5(b) and by matching bit-ratetypes which can be combined with the sampling rate. On the contrary, ina situation of being connected to a security camera or the like suchthat it is necessary to transmit even a change in a fine sound, thereception terminal may determine combination determination informationby adjusting a sampling rate of the NB to the WB, the SWB, or the liketo increase the sampling rate as high as possible and matching bit-ratetypes which can be combined with the sampling rate.

After operation S302, the reception terminal may transmit the determinedcombination determination information to the transmission terminal inoperation S304. Thereafter, when the transmission terminal transmits avoice signal compressed according to the combination determinationinformation, the reception terminal may receive the compressed voicesignal and restore and output a voice signal.

FIG. 8 is a block diagram of an operating apparatus of a receptionterminal for receiving a voice signal, according to an exemplaryembodiment, and the apparatus may include an interface unit 400, acontrol unit 410, and a combination determination unit 420.

Referring to FIG. 8, the interface unit 400 may receive, from atransmission terminal, sampling- and bit-rate request informationincluding sampling list information having at least one sampling typefor determining a sampling rate of a voice signal and bit-rate listinformation having at least one bit-rate type for determining a bit-rateof the voice signal. The interface unit 400 may receive a payload headerinto which the sampling- and bit-rate request information is inserted.The interface unit 400 may receive a payload header including, togetherwith the sampling list information and the bit-rate list information,sampling type confirmation information of a sampling type beingcurrently used, bit-rate type confirmation information of a bit-ratetype being currently used, and the like.

The payload header may be received in a form of a VoIP packet in whichat least one of an RTP header, a UDP header, and an IP header are added.As shown in FIG. 3, the interface unit 400 may receive a VoIP packet inwhich the RTP header, the UDP header, and the IP header are added inaddition to an RTP payload including the payload header, an EVS frame,and padding bits.

The control unit 410 may control the combination determination unit 420to determine a combination of a sampling rate type and a bit rate type,according to the reception of the sampling- and bit-rate requestinformation.

The combination determination unit 420 may determine at least onecombination of sampling types determined from the sampling listinformation and bit-rate types determined from the bit-rate listinformation, according to the sampling- and bit-rate requestinformation.

The combination determination unit 420 may separate the payload header,the RTP header, the UDP header, and the IP header from the received VoIPpacket, extract the sampling list information and the bit-rate listinformation from the separated payload header, and determine the atleast one combination by using the extracted sampling list informationand bit-rate list information.

As shown in FIG. 4(b), the combination determination unit 420 may selectEVS from the sampling rate list information and determine any onesampling rate (for example, 32,000 samples/s). In addition, thecombination determination unit 420 may determine that a bit-rate ofmaximum 24.4 Kbps from the bit-rate list information is used. Each pieceof the combination determination information may indicate (sampling rateidentification information, bit-rate identification information). {(0,0), (1, 1), (1, 2), (2, 3), (2, 4), (2, 5)} as the combinationdetermination information respectively indicate combination informationof sampling rate types and bit-rate types of 7.2 (NB), 8 (WB), 9.6 (WB),13.2 (SWB), 16.4 (SWB), and 24.4 (SWB).

The combination determination unit 420 may determine at least onecombination of sampling rate types and bit-rate types by using at leastone of background noise, audio quality, audio tone of an audio signalincluding the voice signal.

As shown in FIG. 5, a user of the reception terminal may not want tohear high-frequency audio or a sharp sound. In this case, the receptionterminal may analyze received voice frames and determines combinationdetermination information by including a sampling rate which is to bedecreased from the SWB to the WB or NB when an energy ratio ofhigh-frequency components of a specific frequency or higher is aspecific level or higher as shown in FIG. 5(b) and by matching bit-ratetypes which can be combined with the sampling rate. On the contrary, ina situation of being connected to a security camera or the like suchthat it is necessary to transmit even a change in a fine sound, thereception terminal may determine combination determination informationby adjusting a sampling rate of the NB to the WB, the SWB, or the liketo increase the sampling rate as high as possible and matching bit-ratetypes which can be combined with the sampling rate.

The control unit 410 may control the determined combinationdetermination information to be transmitted to the transmissionterminal. According to this, the interface unit 400 may transmit thecombination determination information to the transmission terminal.

Such a negotiation method between a transmission terminal and areception terminal provides maximum flexibility to the transmissionterminal in the selection of a bit rate and a sampling rate to be usedfor a call but has several important restrictions.

First, bit rates used for a call service are influenced by a fare systemand are usually assigned to a set of adjacent bit rates such as 13.2,16.4, and 24.4 Kbps rather than separated to, for example, 7.2, 24.4,and 48 Kbps. That is, bit rates may be assigned in a specific rangeunit, and a bit rate may be adjusted within this range and usedaccording to a transmission condition. In this situation, rather thannegotiating individual bit rates one by one, it may be efficient that arange of a minimum bit rate and a maximum bit rate to be used for a callis negotiated.

FIG. 9 shows a process in which a transmission terminal offers, to areception terminal, bit rates to be respectively used in sending andreceiving directions, by using br-send and br-recv messages, and thereception terminal selects a partial range from among the offered bitrates and answers the selected range.

Herein, the br-send and br-recv messages indicate mutually oppositedirections in view of the transmission terminal and the receptionterminal. In order for a call negotiation to gradually convergeaccording to message exchanges, a bit-rate range of a br-send messagetransmitted by the reception terminal should be a subset of a bit-raterange offered in a br-recv message transmitted by the transmissionterminal. In addition, a bit-rate range of a br-recv message transmittedby the reception terminal should be a subset of a bit-rate range offeredin a br-send message transmitted by the transmission terminal.

Another restriction of the basic negotiation method is a problem that,when a sampling rate has been negotiated but a transmission terminalneeds to adjust the sampling rate due to an unpredicted characteristicof an input signal or background noise, the adjustment should berenegotiated by exchanging messages with a counterpart terminal. If acharacteristic of an input signal or background noise is changed at ahigh speed, even when the transmission terminal continuously triesrenegotiation, the transmission terminal may not meet a characteristicof a currently inputted signal. Even in this case, it may be efficientthat a range of a minimum sampling rate and a maximum sampling rate tobe used for a call is negotiated at once in a call negotiation step.

FIG. 10 shows a process in which a transmission terminal offers, to areception terminal, a range of sampling rates to be respectively used insending and receiving directions, by using bw-send and bw-recv messages,and the reception terminal selects a partial range from among theoffered sampling rates and answers the selected range. Herein, thebw-send and bw-recv messages indicate mutually opposite directions inview of the transmission terminal and the reception terminal. In orderfor a call negotiation to converge, a sampling rate range of a bw-sendmessage transmitted by the reception terminal should be a subset of asampling rate range offered in a bw-recv message transmitted by thetransmission terminal. In addition, a sampling rate range of a bw-recvmessage transmitted by the reception terminal should be a subset of asampling rate range offered in a bw-send message transmitted by thetransmission terminal. As described above, the method of negotiating bitrates and sampling rates in a range unit may be less flexible than themethod of negotiating individual bit rates and individual sampling ratesbut has advantages of meeting a realistic service situation andsimplifying messages to be exchanged. In addition, since both directionsare separately negotiated, even when audio processing capacities oftransmission and reception terminals differ from each other, the methodof negotiating bit rates and sampling rates in a range unit may moreefficiently deal with this situation than the method of negotiatingindividual bit rates and individual sampling rates. When the same bitrate or sampling rate is used for both directions, messages indicatedwith reduced symbols such as br and bw instead of br-send/br-recv andbw-send/bw-recv.

Table 5 includes the definition of these messages. bw, bw-send andbw-recv may be used for negotiation of individual sampling rates such asan NB, a WB, an SWB, and an FB besides sampling rates in a range unit.

TABLE 5 Message Definition br Specifies the range of codec bit-rate tobe used in the session, in kilobits per second, for the sending and thereceiving directions. The parameter can either have: a single bit-rate(br1); or a hyphen-separated pair of two bit-rates (br1-br2). If asingle value is included, this bit-rate, br1, is used. If ahyphen-separated pair of two bit-rates is included, br1 and br2 are usedas the minimum bit-rate and the maximum bit-rate respectively. br1 shallbe smaller than br2. br1 and br2 have a value from the set: 5.9, 7.2, 8,9.6, 13.2, 16.4, 24.4, 32, 48, 64, 96, and 128. 5.9 represents theaverage bit-rate of source controlled variable bit rate (SC-VBR) coding,and 7.2, . . . , 128 represent the bit-rates of constant bit-rate sourcecoding. When the same bit-rate or bit-rate range is defined for thesending and the receiving directions, br should be used but br-send andbr-recv may also be used. br can be used even if the session isnegotiated to be sendonly, recvonly or inactive. For sendonly sessions,br and br-send can be interchangeable used. For recvonly sessions, brand br-recv can be interchangeably used. At least a bandwidth eachnegotiated bit-rate supports shall be included in the negotiatedbandwidth(s). If not present, all bit-rates supporting the negotiatedbandwidth(s) are allowed in the session. When br is not offered for apayload type, the answerer may include br for the payload type in theSDP answer. When br is offered for a payload type and this payload typeis accepted, the answerer shall include br in the SDP answer, and the brshall be a subset of br for the payload type in the SDP offer. br-sendSpecifies the range of codec bit-rate to be used in the session, inkilobits per second, for the sending direction. The parameter can eitherhave: a single bit-rate (br1); or a hyphen-separated pair of twobit-rates (br1-br2). If a single value is included, this bit-rate, br1,is used. If a hyphen-separated pair of two bit-rates is included, br1and br2 are used as the minimum bit-rate and the maximum bit-raterespectively. br1 shall be smaller than br2. br1 and br2 have a valuefrom the set: 5.9, 7.2, 8, 9.6, 13.2, 16.4, 24.4, 32, 48, 64, 96, and128. 5.9 represents the average bit-rate of source controlled variablebit-rate (SC-VBR) coding, and 7.2, . . . , 128 represent the bit-ratesof constant bit-rate source coding. At least a bandwidth each negotiatedbit-rate supports shall be included in the negotiated bandwidth(s). Ifnot present, all bit-rates supporting the negotiated bandwidth(s) areallowed in the session. When br-send is not offered for a payload type,the answerer may include br-recv for the payload type in the SDP answer.When br-send is offered for a payload type and this payload type isaccepted, the answerer shall include br-recv in the SDP answer, and thebr-recv shall be a subset of br-send for the payload type in the SDPoffer. br-recv Specifies the range of codec bit-rate to be used in thesession, in kilobits per second, for the receiving direction. Theparameter can either have: a single bit-rate (br1); or ahyphen-separated pair of two bit-rates (br1-br2). If a single value isincluded, this bit-rate, br1, is used. If a hyphen-separated pair of twobit-rates is included, br1 and br2 are used as the minimum bit-rate andthe maximum bit-rate respectively. br1 shall be smaller than br2. br1and br2 have a value from the set: 5.9, 7.2, 8, 9.6, 13.2, 16.4, 24.4,32, 48, 64, 96, and 128. 5.9 represents the average bit-rate of sourcecontrolled variable bit-rate (SC-VBR) coding, and 7.2, . . . , 128represent the bit-rates of constant bit-rate source coding. At least abandwidth each negotiated bit-rate supports shall be included in thenegotiated bandwidth(s). If not present, all bit-rates supporting thenegotiated bandwidth(s) are allowed in the session. When br-recv is notoffered for a payload type, the answerer may include br-send for thepayload type in the SDP answer. When br-recv is offered for a payloadtype and this payload type is accepted, the answerer shall includebr-send in the SDP answer, and the br-send shall be a subset of br-recvfor the payload type in the SDP offer. bw Specifies the bandwidth to beused in the session for the sending and the receiving directions. bw hasa value from the set: nb, wb, swb, fb, nb-wb, nb-swb, and nb-fb. nb, wb,swb, and fb represent narrowband, wideband, super-wideband, and fullbandrespectively, and nb-wb, nb-swb, and nb-fb represent all bandwidths fromnarrowband to wideband, super-wideband, and fullband respectively. Whenthe same bandwidth or bandwidth range is defined for the sending and thereceiving directions, bw should be used but bw-send and bw-recv may alsobe used. bw can be used even if the session is negotiated to besendonly, recvonly or inactive. For sendonly session, bw and bw-send canbe interchangeable used. For recvonly sessions, bw and bw-recv can beinterchangeably used. If not present, all bandwidths the negotiatedbit-rate(s) support are allowed in the session. When bw is not offeredfor a payload type, the answerer may include bw for the payload type inthe SDP answer. When bw is offered for a payload type and this payloadtype is accepted, the answerer shall include bw in the SDP answer, andthe bw shall be a subset of bw for the payload type in the SDP offer.)bw-send Specifies the bandwidth to be used in the session for thesending direction. bw-send has a value from the set: nb, wb, swb, fb,nb-wb, nb-swb, and nb-fb. nb, wb, swb, and fb represent narrowband,wideband, super-wideband, and fullband respectively, and nb-wb, nb-swb,and nb-fb represent all bandwidths from narrowband to wideband,super-wideband, and fullband respectively. At least a negotiatedbit-rate shall support each negotiated bandwidth. If not present, allbandwidths the negotiated bit-rate(s) support are allowed in thesession. When bw-send is not offered for a payload type, the answerermay include bw-recv for the payload type in the SDP answer. When bw-sendis offered for a payload type and this payload is accepted, the answerershall include bw-recv in the SDP answer, and the bw-recv shall be asubset of bw-send for the payload type in the SDP offer. bw-recvSpecifies the bandwidth to be used in the session for the receivingdirection, bw-recv has a value from the set: nb, wb, swb, fb, nb-wb,nb-swb, and nb-fb. nb, wb, swb, and fb represent narrowband, wideband,super-wideband, and fullband respectively, and nb-wb, nb-swb, and nb-fbrepresent all bandwidths from narrowband to wideband, super-wideband,and fullband respectively. At least a negotiated bit-rate shall supporteach negotiated bandwidth. If not present, all bandwidths the negotiatedbit-rate(s) support are allowed in the session. When bw-recv is notoffered for a payload type, the answerer may include bw-send for thepayload type in the SDP answer. When bw-recv is offered for a payloadtype and this payload is accepted, the answerer shall include bw-send inthe SDP answer, and the bw-send shall be a subset of bw-recv for thepayload type in the SDP offer.

In FIG. 11, a transmission terminal offers, to a reception terminal,that an EVS codec uses a bit-rate in a range of 5.9 to 64 Kbps, and thereception terminal adjusts upper and lower limits of the range, modifiesthe range such that a bit-rate in a range of 13.2 to 24.4 Kbps is usedfor a call, and answers the modified range back to the transmissionterminal.

In FIG. 12, a transmission terminal offers, to a reception terminal,that an EVS codec uses a sampling rate in a range of the NB to the SWB,and the reception terminal adjusts upper and lower limits of the range,modifies the range such that a sampling rate in a range of the NB to theWB is used, and answers the modified range back to the transmissionterminal.

In FIG. 13, a transmission terminal offers, to a reception terminal,that an EVS codec uses a bit-rate in a range of 5.9 to 64 Kbps and asampling rate in a range of the NB to the SWB, and the receptionterminal adjusts upper and lower limits of the bit-rate range and anupper limit of the sampling rate range, modifies the ranges such that abit-rate in a range of 13.2 to 24.4 Kbps and a sampling rate in a rangeof the NB to the WB are used, and answers the modified ranges back tothe transmission terminal.

In FIG. 14, a transmission terminal offers, to a reception terminal,that an EVS codec uses a bit-rate in a range of 5.9 to 24.4 Kbps and asampling rate in a range of the NB to the SWB, and the receptionterminal modifies the ranges such that a bit-rate of 13.2 Kbps and asampling rate in a range of the NB to the WB are used in a receivingdirection and a bit-rate in the range of 5.9 to 24.4 Kbps and a samplingrate in the range of the NB to the SWB are used as requested in asending direction, and answers the modified ranges back to thetransmission terminal.

The methods according to exemplary embodiments may be implemented ashardware or software or as a combination of hardware and software. Whenthe methods are implemented as software, a computer-readable storagemedium for storing one or more programs (software modules) may beprovided. The one or more programs stored in the computer-readablestorage medium are configured for execution by one or more processors inan electronic device. The one or more programs include instructions ofcommanding the electronic device to execute the methods according toembodiments.

These programs (software modules or software) may be stored in anonvolatile memory including random access memory (RAM) and flashmemory, read-only memory (ROM), electrically erasable programmable ROM(EEPROM), a magnetic disc storage device, compact disc-ROM (CD-ROM), adigital versatile disc (DVD), another type of optical storage device, ora magnetic cassette. Alternatively, the programs may be stored in amemory including some or all thereof. In addition, each of the memoriesdescribed above may be plural in number.

Alternatively, the programs may be stored in an attachable storagedevice which can access the electronic device through a communicationnetwork including the Internet, an intranet, a local area network (LAN),a wide LAN (WLAN), and a storage area network (SAN), taken alone or incombination. This storage device may access the electronic devicethrough an external port. Alternatively, a separate storage device onthe communication network may access a portable electronic device.

The invention claimed is:
 1. An operating method of a transmissionterminal for transmitting an audio signal, the method comprising:generating first bit rate information for a sending direction of thetransmission terminal indicating a first bit rate range, and second bitrate information for a receiving direction of the transmission terminalindicating a second bit rate range; transmitting the first bit rateinformation and the second bit rate information to a reception terminal;receiving combination determination information from the receptionterminal; compressing the audio signal according to the receivedcombination determination information; and transmitting the compressedaudio signal to the reception terminal, wherein the combinationdetermination information is determined based on third bit rateinformation for a receiving direction of the reception terminalindicating a third bit rate range, by the reception terminal, andwherein the third bit rate range is comprised in the first bit raterange.
 2. The method of claim 1, wherein the combination determinationinformation is adjusted during a call by the reception terminal and thecombination determination information is received during a call by thetransmission terminal.
 3. The method of claim 1, wherein the combinationdetermination information is determined by using at least one ofbackground noise, audio quality, and audio tone of the audio signal. 4.The method of claim 1, wherein when the first bit rate range and thesecond bit rate range are different each other, the first bit rateinformation and the second bit rate information are transmitted to thereception terminal through different messages.
 5. The method of claim 1,wherein when the first bit rate range and the second bit rate range arethe same, the first bit rate information and the second bit rateinformation are transmitted to the reception terminal through a singlemessage.
 6. An operating method of a reception terminal for receiving anaudio signal, the method comprising: receiving, from a transmissionterminal, first bit rate information for a sending direction of thetransmission terminal indicating a first bit rate range, and second bitrate information for a receiving direction of the transmission terminalindicating a second bit rate range; determining combinationdetermination information based on third bit rate information for areceiving direction of the reception terminal indicating a third bitrate range; and transmitting the combination determination informationto the transmission terminal, wherein the third bit rate range iscomprised in the first bit rate range.
 7. The method of claim 6, whereinthe combination determination information is adjusted during a call bythe reception terminal and the combination determination information isreceived during a call by the transmission terminal.
 8. The method ofclaim 6, wherein the combination determination information is determinedby using at least one of background noise, audio quality, and audio toneof the audio signal.